Refactor webrtc/{api,audio} and modules/audio_coding for GN check

webrtc/modules/audio_coding/BUILD.gn

 # Copyright (c) 2014 The WebRTC project authors. All Rights Reserved.
 #
 # Use of this source code is governed by a BSD-style license
 # that can be found in the LICENSE file in the root of the source
 # tree. An additional intellectual property rights grant can be found
 # in the file PATENTS.  All contributing project authors may
 # be found in the AUTHORS file in the root of the source tree.
 
 import("../../build/webrtc.gni")
 import("audio_coding.gni")
@@ skipping 59 matching lines @@
     ":audio_format",
 
     # TODO(charujain): Clean this dependency when downstream projects are
     # updated to properly depend on audio_format_conversion target.
     ":audio_format_conversion",
     "../../base:rtc_base_approved",
   ]
 }
 
 rtc_static_library("builtin_audio_decoder_factory") {
+  # TODO(kjellander): Remove (bugs.webrtc.org/6828)
+  # Errors on cyclic dependency with :isac_fix if enabled.
+  check_includes = false
+
   sources = [
     "codecs/builtin_audio_decoder_factory.cc",
     "codecs/builtin_audio_decoder_factory.h",
   ]
   deps = [
            "../..:webrtc_common",
+           "../../base:rtc_base_approved",
            ":audio_decoder_factory_interface",
          ] + audio_codec_deps
   defines = audio_codec_defines
 }
 
 rtc_static_library("rent_a_codec") {
+  # TODO(kjellander): Remove (bugs.webrtc.org/6828)
+  # Gives cyclic dependency with :neteq and :audio_coding if enabled.
+  check_includes = false
+
   sources = [
     "acm2/acm_codec_database.cc",
     "acm2/acm_codec_database.h",
     "acm2/rent_a_codec.cc",
     "acm2/rent_a_codec.h",
   ]
-  deps = [ "../..:webrtc_common" ] + audio_codec_deps
+  deps = [
+           ":audio_decoder_interface",
+           "../..:webrtc_common",
+           "../../base:rtc_base_approved",
+         ] + audio_codec_deps
   defines = audio_codec_defines
 }
 
 config("audio_coding_config") {
   include_dirs = [
     "include",
     "../include",
   ]
 }
 
@@ skipping 24 matching lines @@
   }
 
   if (is_win) {
     cflags = [
       # TODO(kjellander): Bug 261: fix this warning.
       "/wd4373",  # virtual function override.
     ]
   }
 
   deps = audio_coding_deps + [
+           ":audio_decoder_interface",
+           ":audio_decoder_factory_interface",
+           ":builtin_audio_decoder_factory",
            ":neteq",
            ":rent_a_codec",
+           "../../base:rtc_base_approved",
            "../../logging:rtc_event_log_api",
          ]
   defines = audio_coding_defines
 }
 
 rtc_static_library("audio_decoder_interface") {
   sources = [
     "codecs/audio_decoder.cc",
     "codecs/audio_decoder.h",
     "codecs/legacy_encoded_audio_frame.cc",
@@ skipping 28 matching lines @@
     "codecs/cng/audio_encoder_cng.cc",
     "codecs/cng/audio_encoder_cng.h",
     "codecs/cng/webrtc_cng.cc",
     "codecs/cng/webrtc_cng.h",
   ]
 
   public_configs = [ ":cng_config" ]
 
   deps = [
     ":audio_encoder_interface",
+    "../..:webrtc_common",
+    "../../base:rtc_base_approved",
     "../../common_audio",
   ]
 }
 
 config("red_config") {
   include_dirs = [ "codecs/red" ]
 }
 
 rtc_static_library("red") {
   sources = [
     "codecs/red/audio_encoder_copy_red.cc",
     "codecs/red/audio_encoder_copy_red.h",
   ]
 
   public_configs = [ ":red_config" ]
 
   deps = [
     ":audio_encoder_interface",
+    "../../base:rtc_base_approved",
     "../../common_audio",
   ]
 }
 
 config("g711_config") {
   include_dirs = [
     "../../..",
     "codecs/g711/include",
   ]
 }
 
 rtc_static_library("g711") {
   sources = [
     "codecs/g711/audio_decoder_pcm.cc",
     "codecs/g711/audio_decoder_pcm.h",
     "codecs/g711/audio_encoder_pcm.cc",
     "codecs/g711/audio_encoder_pcm.h",
     "codecs/g711/g711.c",
     "codecs/g711/g711.h",
     "codecs/g711/g711_interface.c",
     "codecs/g711/g711_interface.h",
   ]
 
   public_configs = [ ":g711_config" ]
 
   deps = [
     ":audio_decoder_interface",
     ":audio_encoder_interface",
+    "../..:webrtc_common",
+    "../../base:rtc_base_approved",
   ]
 }
 
 config("g722_config") {
   include_dirs = [
     "../../..",
     "codecs/g722/include",
   ]
 }
 
 rtc_static_library("g722") {
   sources = [
     "codecs/g722/audio_decoder_g722.cc",
     "codecs/g722/audio_decoder_g722.h",
     "codecs/g722/audio_encoder_g722.cc",
     "codecs/g722/audio_encoder_g722.h",
     "codecs/g722/g722_decode.c",
     "codecs/g722/g722_enc_dec.h",
     "codecs/g722/g722_encode.c",
     "codecs/g722/g722_interface.c",
     "codecs/g722/g722_interface.h",
   ]
 
   public_configs = [ ":g722_config" ]
 
   deps = [
     ":audio_decoder_interface",
     ":audio_encoder_interface",
+    "../..:webrtc_common",
+    "../../base:rtc_base_approved",
   ]
 }
 
 config("ilbc_config") {
   include_dirs = [
     "../../..",
     "codecs/ilbc/include",
   ]
 }
 
@@ skipping 142 matching lines @@
     "codecs/ilbc/window32_w32.h",
     "codecs/ilbc/xcorr_coef.c",
     "codecs/ilbc/xcorr_coef.h",
   ]
 
   public_configs = [ ":ilbc_config" ]
 
   deps = [
     ":audio_decoder_interface",
     ":audio_encoder_interface",
+    "../..:webrtc_common",
     "../../base:rtc_base_approved",
     "../../common_audio",
   ]
 }
 
 rtc_static_library("isac_common") {
+  # TODO(kjellander): Remove (bugs.webrtc.org/6828)
+  # Has a cyclic dependency with :isac if checks are enabled.
+  check_includes = false
+
   sources = [
     "codecs/isac/audio_encoder_isac_t.h",
     "codecs/isac/audio_encoder_isac_t_impl.h",
     "codecs/isac/locked_bandwidth_info.cc",
     "codecs/isac/locked_bandwidth_info.h",
   ]
+  deps = [
+    ":audio_encoder_interface",
+    "../../base:rtc_base_approved",
+  ]
 }
 
 config("isac_config") {
   include_dirs = [
     "../../..",
     "codecs/isac/main/include",
   ]
 }
 
 rtc_static_library("isac") {
@@ skipping 127 matching lines @@
     "codecs/isac/fix/source/structs.h",
     "codecs/isac/fix/source/transform.c",
     "codecs/isac/fix/source/transform_tables.c",
   ]
 
   public_configs = [ ":isac_fix_config" ]
 
   deps = [
     ":audio_decoder_interface",
     ":audio_encoder_interface",
+    ":builtin_audio_decoder_factory",
     ":isac_common",
+    "../..:webrtc_common",
+    "../../base:rtc_base_approved",
     "../../common_audio",
     "../../system_wrappers",
   ]
 
   if (rtc_build_with_neon) {
     deps += [ ":isac_neon" ]
   }
 
   if (current_cpu == "arm" && arm_version >= 7) {
     sources += [
@@ skipping 78 matching lines @@
     "codecs/pcm16b/audio_encoder_pcm16b.cc",
     "codecs/pcm16b/audio_encoder_pcm16b.h",
     "codecs/pcm16b/pcm16b.c",
     "codecs/pcm16b/pcm16b.h",
   ]
 
   deps = [
     ":audio_decoder_interface",
     ":audio_encoder_interface",
     ":g711",
+    "../..:webrtc_common",
+    "../../base:rtc_base_approved",
   ]
 
   public_configs = [ ":pcm16b_config" ]
 }
 
 config("opus_config") {
   include_dirs = [ "../../.." ]
 }
 
 rtc_static_library("webrtc_opus") {
   sources = [
     "codecs/opus/audio_decoder_opus.cc",
     "codecs/opus/audio_decoder_opus.h",
     "codecs/opus/audio_encoder_opus.cc",
     "codecs/opus/audio_encoder_opus.h",
     "codecs/opus/opus_inst.h",
     "codecs/opus/opus_interface.c",
     "codecs/opus/opus_interface.h",
   ]
 
   deps = [
     ":audio_decoder_interface",
     ":audio_encoder_interface",
     ":audio_network_adaptor",
+    "../..:webrtc_common",
     "../../base:rtc_analytics",
     "../../base:rtc_base_approved",
+    "../../system_wrappers",
   ]
 
   defines = []
   if (rtc_opus_variable_complexity) {
     defines += [ "WEBRTC_OPUS_VARIABLE_COMPLEXITY=1" ]
   } else {
     defines += [ "WEBRTC_OPUS_VARIABLE_COMPLEXITY=0" ]
   }
 
   if (rtc_build_opus) {
@@ skipping 39 matching lines @@
     "audio_network_adaptor/dtx_controller.h",
     "audio_network_adaptor/fec_controller.cc",
     "audio_network_adaptor/fec_controller.h",
     "audio_network_adaptor/frame_length_controller.cc",
     "audio_network_adaptor/frame_length_controller.h",
     "audio_network_adaptor/include/audio_network_adaptor.h",
   ]
 
   deps = [
     "../..:webrtc_common",
+    "../../base:rtc_base_approved",
+    "../../common_audio",
     "../../system_wrappers",
   ]
 
   if (rtc_enable_protobuf) {
     deps += [
       ":ana_config_proto",
       ":ana_debug_dump_proto",
     ]
     defines = [ "WEBRTC_AUDIO_NETWORK_ADAPTOR_DEBUG_DUMP" ]
   }
 }
 
 config("neteq_config") {
   include_dirs = [
     # Need Opus header files for the audio classifier.
     "//third_party/opus/src/celt",
     "//third_party/opus/src/src",
   ]
 }
 
 rtc_static_library("neteq") {
+  # TODO(kjellander): Remove (bugs.webrtc.org/6828)
+  # Cyclic dependency with :audio_coding if enabled.
+  check_includes = false
+
   sources = [
     "neteq/accelerate.cc",
     "neteq/accelerate.h",
     "neteq/audio_classifier.cc",
     "neteq/audio_classifier.h",
     "neteq/audio_decoder_impl.cc",
     "neteq/audio_decoder_impl.h",
     "neteq/audio_multi_vector.cc",
     "neteq/audio_multi_vector.h",
     "neteq/audio_vector.cc",
@@ skipping 59 matching lines @@
     "neteq/tick_timer.h",
     "neteq/time_stretch.cc",
     "neteq/time_stretch.h",
     "neteq/timestamp_scaler.cc",
     "neteq/timestamp_scaler.h",
   ]
 
   public_configs = [ ":neteq_config" ]
 
   deps = [
+    ":audio_decoder_factory_interface",
     ":audio_decoder_interface",
+    ":audio_format",
     ":builtin_audio_decoder_factory",
     ":cng",
     ":g711",
+    ":isac_fix",
     ":pcm16b",
     ":rent_a_codec",
     "../..:webrtc_common",
+    "../../base:gtest_prod",
+    "../../base:rtc_base_approved",
     "../../common_audio",
     "../../system_wrappers",
   ]
 
   defines = []
 
   if (rtc_include_ilbc) {
     defines += [ "WEBRTC_CODEC_ILBC" ]
     deps += [ ":ilbc" ]
   }
@@ skipping 13 matching lines @@
     deps += [ ":g722" ]
   }
 }
 
 # Although providing only test support, this target must be outside of the
 # rtc_include_tests conditional. The reason is that it supports fuzzer tests
 # that ultimately are built and run as a part of the Chromium ecosystem, which
 # does not set the rtc_include_tests flag.
 rtc_source_set("neteq_test_minimal") {
   testonly = true
+
+  # TODO(kjellander): Remove (bugs.webrtc.org/6828)
+  # Has cyclic dependency with :neteq_unittest_tools
+  check_includes = false
+
   sources = [
     "neteq/tools/encode_neteq_input.cc",
     "neteq/tools/encode_neteq_input.h",
     "neteq/tools/neteq_test.cc",
     "neteq/tools/neteq_test.h",
   ]
 
   if (!build_with_chromium && is_clang) {
     # Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163).
     suppressed_configs += [ "//build/config/clang:find_bad_constructs" ]
   }
+
+  deps = [
+    ":audio_encoder_interface",
+    ":builtin_audio_decoder_factory",
+    ":neteq",
+    "../..:webrtc_common",
+    "../../base:rtc_base_approved",
+  ]
 }
 
 if (rtc_include_tests) {
   group("audio_coding_tests") {
     testonly = true
     public_deps = [
       ":RTPchange",
       ":RTPencode",
       ":RTPjitter",
       ":RTPtimeshift",
@@ skipping 30 matching lines @@
     testonly = true
     sources = [
       "acm2/acm_receive_test.cc",
       "acm2/acm_receive_test.h",
     ]
 
     defines = audio_coding_defines
 
     deps = audio_coding_deps + [
              ":audio_coding",
+             ":audio_format_conversion",
+             ":audio_decoder_factory_interface",
+             ":builtin_audio_decoder_factory",
              ":neteq_unittest_tools",
+             "../../base:rtc_base_approved",
+             "../../test:test_support",
              "//testing/gtest",
            ]
   }
 
   rtc_source_set("acm_send_test") {
     testonly = true
     sources = [
       "acm2/acm_send_test.cc",
       "acm2/acm_send_test.h",
     ]
 
     defines = audio_coding_defines
 
     deps = audio_coding_deps + [
              ":audio_coding",
+             ":audio_decoder_interface",
+             ":audio_encoder_interface",
              ":neteq_unittest_tools",
+             "../../base:rtc_base_approved",
+             "../../test:test_support",
              "//testing/gtest",
            ]
   }
 
   rtc_executable("delay_test") {
     testonly = true
     sources = [
       "test/Channel.cc",
       "test/PCMFile.cc",
       "test/delay_test.cc",
       "test/utility.cc",
     ]
 
     deps = [
       ":audio_coding",
       ":audio_format_conversion",
       "../../:webrtc_common",
+      "../../base:rtc_base_approved",
       "../../system_wrappers",
       "../../system_wrappers:system_wrappers_default",
       "../../test:test_support",
       "../rtp_rtcp",
       "//testing/gtest",
       "//third_party/gflags:gflags",
     ]
   }  # delay_test
 
   rtc_executable("insert_packet_with_timing") {
     testonly = true
     sources = [
       "test/Channel.cc",
       "test/PCMFile.cc",
       "test/insert_packet_with_timing.cc",
     ]
 
     if (!build_with_chromium && is_clang) {
       # Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163).
       suppressed_configs += [ "//build/config/clang:find_bad_constructs" ]
     }
 
     deps = [
       ":audio_coding",
       ":audio_format_conversion",
       "../../:webrtc_common",
+      "../../base:rtc_base_approved",
       "../../system_wrappers",
       "../../system_wrappers:system_wrappers_default",
       "../../test:test_support",
       "../rtp_rtcp",
       "//testing/gtest",
       "//third_party/gflags:gflags",
     ]
   }  # insert_packet_with_timing
 
   audio_decoder_unittests_resources =
@@ skipping 50 matching lines @@
   if (rtc_enable_protobuf) {
     proto_library("neteq_unittest_proto") {
       sources = [
         "neteq/neteq_unittest.proto",
       ]
       proto_out_dir = "webrtc/audio_coding/neteq"
     }
 
     rtc_static_library("rtc_event_log_source") {
       testonly = true
+
+      # TODO(kjellander): Remove (bugs.webrtc.org/6828)
+      # Needs call.h to be moved to webrtc/api first.
+      check_includes = false
+
       sources = [
         "neteq/tools/rtc_event_log_source.cc",
         "neteq/tools/rtc_event_log_source.h",
       ]
 
       if (!build_with_chromium && is_clang) {
         # Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163).
         suppressed_configs += [ "//build/config/clang:find_bad_constructs" ]
       }
 
       deps = [
+        "../../base:rtc_base_approved",
         "../../logging:rtc_event_log_parser",
       ]
       public_deps = [
         "../../logging:rtc_event_log_proto",
       ]
     }
 
     rtc_test("neteq_rtpplay") {
       testonly = true
       defines = []
@@ skipping 10 matching lines @@
       if (is_win) {
         cflags = [
           # TODO(kjellander): bugs.webrtc.org/261: Fix this warning.
           "/wd4373",  # virtual function override.
         ]
       }
 
       deps += [
         ":neteq",
         ":neteq_unittest_tools",
+        "../..:webrtc_common",
+        "../../base:rtc_base_approved",
         "../../system_wrappers:system_wrappers_default",
         "../../test:test_support",
         "//third_party/gflags",
       ]
     }
   }
 
   rtc_test("audio_codec_speed_tests") {
     testonly = true
     defines = []
@@ skipping 11 matching lines @@
     }
 
     if (is_android) {
       deps += [ "//testing/android/native_test:native_test_native_code" ]
       shard_timeout = 900
     }
 
     deps += [
       ":isac_fix",
       ":webrtc_opus",
+      "../..:webrtc_common",
+      "../../base:rtc_base_approved",
       "../../system_wrappers:system_wrappers_default",
       "../../test:test_main",
       "../audio_processing",
       "//testing/gtest",
     ]
   }
 
   rtc_source_set("neteq_test_support") {
     testonly = true
     sources = [
       "neteq/tools/neteq_external_decoder_test.cc",
       "neteq/tools/neteq_external_decoder_test.h",
       "neteq/tools/neteq_performance_test.cc",
       "neteq/tools/neteq_performance_test.h",
     ]
 
     if (!build_with_chromium && is_clang) {
       # Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163).
       suppressed_configs += [ "//build/config/clang:find_bad_constructs" ]
     }
 
     deps = [
+      ":audio_decoder_interface",
+      ":builtin_audio_decoder_factory",
       ":neteq",
       ":neteq_unittest_tools",
       ":pcm16b",
+      "../..:webrtc_common",
+      "../../base:rtc_base_approved",
+      "../../system_wrappers",
+      "../../test:test_support",
       "//testing/gtest",
     ]
   }
 
   rtc_source_set("neteq_quality_test_support") {
     testonly = true
     sources = [
       "neteq/tools/neteq_quality_test.cc",
       "neteq/tools/neteq_quality_test.h",
     ]
 
     if (!build_with_chromium && is_clang) {
       # Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163).
       suppressed_configs += [ "//build/config/clang:find_bad_constructs" ]
     }
 
     deps = [
+      ":builtin_audio_decoder_factory",
       ":neteq",
       ":neteq_unittest_tools",
+      "../..:webrtc_common",
+      "../../base:rtc_base_approved",
+      "../../test:test_support",
       "//testing/gtest",
       "//third_party/gflags",
     ]
   }
 
   config("neteq_unittest_tools_config") {
     include_dirs = [ "tools" ]
   }
 
   rtc_source_set("neteq_unittest_tools") {
@@ skipping 30 matching lines @@
     ]
 
     public_configs = [ ":neteq_unittest_tools_config" ]
 
     if (!build_with_chromium && is_clang) {
       # Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163).
       suppressed_configs += [ "//build/config/clang:find_bad_constructs" ]
     }
 
     deps = [
-      ":neteq_test_minimal",
+      ":audio_decoder_interface",
+      ":audio_encoder_interface",
+      ":pcm16b",
+      "../..:webrtc_common",
+      "../../base:rtc_base_approved",
       "../../common_audio",
       "../../test:rtp_test_utils",
       "../rtp_rtcp",
     ]
 
+    public_deps = [
+      ":neteq_test_minimal",
+    ]
+
     if (rtc_enable_protobuf) {
       deps += [ ":rtc_event_log_source" ]
     }
   }
 
   rtc_source_set("neteq_test_tools") {
     testonly = true
     sources = [
       "neteq/test/NETEQTEST_DummyRTPpacket.cc",
       "neteq/test/NETEQTEST_DummyRTPpacket.h",
@@ skipping 49 matching lines @@
   rtc_executable("RTPencode") {
     testonly = true
 
     deps = [
       # TODO(hlundin): Make RTPencode use ACM to encode files.
       ":cng",
       ":g711",
       ":g722",
       ":ilbc",
       ":isac",
+      ":neteq",
       ":neteq_test_tools",
       ":pcm16b",
       ":webrtc_opus",
+      "../..:webrtc_common",
+      "../../base:rtc_base_approved",
       "../../common_audio",
     ]
 
     configs += [ ":RTPencode_config" ]
 
     sources = [
       "neteq/test/RTPencode.cc",
     ]
 
     include_dirs = [
@@ skipping 22 matching lines @@
   }
 
   rtc_executable("rtpcat") {
     testonly = true
 
     sources = [
       "neteq/tools/rtpcat.cc",
     ]
 
     deps = [
+      "../../base:rtc_base_approved",
       "../../system_wrappers:system_wrappers_default",
       "../../test:rtp_test_utils",
       "//testing/gtest",
     ]
   }
 
   rtc_executable("RTPtimeshift") {
     testonly = true
 
     sources = [
       "neteq/test/RTPtimeshift.cc",
     ]
 
     deps = [
       ":neteq_test_tools",
+      "../../test:test_support",
       "//testing/gtest",
     ]
   }
 
   rtc_executable("RTPjitter") {
     testonly = true
     deps = [
       "../..:webrtc_common",
       "../../base:rtc_base_approved",
+      "../../test:test_support",
       "//testing/gtest",
     ]
     sources = [
       "neteq/test/RTPjitter.cc",
     ]
   }
 
   rtc_executable("rtp_analyze") {
     testonly = true
 
@@ skipping 37 matching lines @@
   rtc_executable("neteq_speed_test") {
     testonly = true
 
     sources = [
       "neteq/test/neteq_speed_test.cc",
     ]
 
     deps = [
       ":neteq",
       ":neteq_test_support",
+      "../..:webrtc_common",
       "../../system_wrappers:system_wrappers_default",
       "../../test:test_support",
       "//third_party/gflags",
     ]
   }
 
   rtc_executable("audio_classifier_test") {
     testonly = true
     sources = [
       "neteq/test/audio_classifier_test.cc",
@@ skipping 10 matching lines @@
 
     sources = [
       "neteq/test/neteq_ilbc_quality_test.cc",
     ]
 
     deps = [
       ":ilbc",
       ":neteq",
       ":neteq_quality_test_support",
       ":neteq_unittest_tools",
+      "../..:webrtc_common",
+      "../../base:rtc_base_approved",
       "../../system_wrappers:system_wrappers_default",
       "../../test:test_main",
       "//testing/gtest",
       "//third_party/gflags",
     ]
   }
 
   rtc_executable("neteq_isac_quality_test") {
     testonly = true
 
     sources = [
       "neteq/test/neteq_isac_quality_test.cc",
     ]
 
     deps = [
       ":isac_fix",
       ":neteq",
       ":neteq_quality_test_support",
+      "../../base:rtc_base_approved",
       "../../test:test_main",
       "//testing/gtest",
       "//third_party/gflags",
     ]
   }
 
   rtc_executable("neteq_pcmu_quality_test") {
     testonly = true
 
     sources = [
       "neteq/test/neteq_pcmu_quality_test.cc",
     ]
 
     deps = [
       ":g711",
       ":neteq",
       ":neteq_quality_test_support",
+      "../../base:rtc_base_approved",
       "../../test:test_main",
       "//testing/gtest",
       "//third_party/gflags",
     ]
   }
 
   rtc_executable("isac_fix_test") {
     testonly = true
 
     sources = [
@@ skipping 127 matching lines @@
 
     sources = [
       "codecs/opus/opus_fec_test.cc",
     ]
 
     deps = [
       ":webrtc_opus",
       "../../base:rtc_base_approved",
       "../../common_audio",
       "../../test:test_main",
-      "../../test:test_support",
       "//testing/gtest",
     ]
 
     if (!build_with_chromium && is_clang) {
       # Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163).
       suppressed_configs += [ "//build/config/clang:find_bad_constructs" ]
     }
   }
 }